Re: [Asterisk-Users] gsm picocells
James Harper wrote: I believe the OP wants to use GSM handsets as extensions, like running your own localized GSM network. That's not the same as using a GSM terminal to connect Asterisk to the cellular network. Correct! IP Access makes such products. http://www.ipaccess.com/products/nanoBTS.htm That looks about right. All problems of spectrum licensing etc aside, In most countries, operating RF equipment in the 900/180MHz GSM bands require licensing. Alcatel and Siemens both have picocell support for their PBX but I have never seen one of those in opertion. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
It seems the proxy address is added to all incoming calls to the Cisco phone. On 3/16/06, Tim Connolly [EMAIL PROTECTED] wrote: I'm not sure this is the issue. Every call seem to get the proxy address added whether it's the main proxy or the backup. What has to match to make the phone NOT append the proxy address? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Wednesday, March 15, 2006 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? That's probably what is happening on my end. Any suggestions on how to fix this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, March 14, 2006 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We only had the problem when the call was redirected from one server to another. So if a phone was called from another phone on the server, the called worked perfectly, but if it was redirected from another server, we got the proxy added to the end. Doesn't help when you're trying to make the existence of multiple servers transparent. Aaron Chris Stenton wrote: Maybe I have something strange in my dial plan but I have no problem just hitting dial from missed calls under 8.2. Chris - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 8:44 PM Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We rolled back to 7.4 cause of that too. 7.5 has a strange bug where if the server loses connection, the phone's just don't try re-registering. Aaron Tim Connolly wrote: Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Server freeze with meetme and sip GSM users
Hi Brent Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back conf-onlyperson. This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if I hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM Enconding problem as I suspected first, this happens with every encoding. magma*CLI -- Executing Answer(SIP/11-9d7c, ) in new stack -- Executing MeetMe(SIP/11-9d7c, 555) in new stack -- Created MeetMe conference 1023 for conference '555' -- Playing 'conf-onlyperson' (language 'de') magma*CLI Freeze! Any other who can reproduce that freeze? Kernel 2.6.15 / * 1.2.5 / ztdummy 1.2.4 -Benoit- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm picocells
On Sat, Mar 18, 2006 at 04:49:53PM +1100, James Harper wrote: I believe the OP wants to use GSM handsets as extensions, like running your own localized GSM network. That's not the same as using a GSM terminal to connect Asterisk to the cellular network. Correct! IP Access makes such products. http://www.ipaccess.com/products/nanoBTS.htm That looks about right. All problems of spectrum licensing etc aside, the product claims to use Ethernet as the wired access medium, but appears to need to connect to a much meatier box as part of a packaged solution. The site doesn't seem to give much away, including price. That's the trouble with GSM, the cell (or picocell) is just part of the infrastructure required. A cell is actually a BSC (basetation controller). BSC's are controlled by MSC's (Mobile switching centre), an MSC will control multiple BSCs and MSC talk to each other. We're in SS7 land now. You also need an HLR (home location register), SMSC (if you want your users to do SMS) and then all the GPRS bits for MMS/data/etc. IP.Access's picocell uses IP backhaul so can be deployed easily in remote sites. They cost around GBP 2,000. In the UK there are between 7 and 12 low power GSM national licenses becoming available (in the old GSM/DECT guard bands). Need to get you intent to bid into Ofcom (and payment) on the 21st b/n 10am a 5.30pm. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] gsm picocells
I believe the OP wants to use GSM handsets as extensions, like running your own localized GSM network. That's not the same as using a GSM terminal to connect Asterisk to the cellular network. Correct! IP Access makes such products. http://www.ipaccess.com/products/nanoBTS.htm That looks about right. All problems of spectrum licensing etc aside, the product claims to use Ethernet as the wired access medium, but appears to need to connect to a much meatier box as part of a packaged solution. The site doesn't seem to give much away, including price. That's the trouble with GSM, the cell (or picocell) is just part of the infrastructure required. A cell is actually a BSC (basetation controller). BSC's are controlled by MSC's (Mobile switching centre), an MSC will control multiple BSCs and MSC talk to each other. We're in SS7 land now. You also need an HLR (home location register), SMSC (if you want your users to do SMS) and then all the GPRS bits for MMS/data/etc. IP.Access's picocell uses IP backhaul so can be deployed easily in remote sites. They cost around GBP 2,000. Ah. More complicated than I'd hoped but not more than I suspected :) So the product that can accept gsm phone registrations and calls and trunk them to asterisk via E1/TDMoE/TDMoIP/SIP/IAX is still wishware? Oh well. I guess hybrid gsm/dect/wifi phones will reach maturity first which is probably a better solution to the problem anyway. Thanks for the info, if nothing else I'm now a little wiser on the subject. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fake Ring Tone/Compile Addon
Subject: Re: [Asterisk-Users] Fake Ring Tone/Compile AddonTo: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowed Kenige Ho wrote: Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem?Don't use the fake ring option to dial. This is the r option. Dear Manxpower, I didn't use the 'r' option in my Dial command, and the funny thing that out going to my SIP Phones doesn't have fake ring tone. But there is always fake ring tone, when sending out to the VoIP Gateway and I am sure that I don't set it in the VoIP gateway. Please help. Thanks. Regards, Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm picocells
On Sat, Mar 18, 2006 at 10:16:27PM +1100, James Harper wrote: [snip] Ah. More complicated than I'd hoped but not more than I suspected :) So the product that can accept gsm phone registrations and calls and trunk them to asterisk via E1/TDMoE/TDMoIP/SIP/IAX is still wishware? Oh well. I guess hybrid gsm/dect/wifi phones will reach maturity first which is probably a better solution to the problem anyway. There IS an initiative called UMA (unlicensed mobiel access) whereby a GSM phone can roam on to a local WiFi or Bluetooth network, the specs are freely published. In the UK BT are offering a service based on this called Fusion, which uses a Bluetooth basestation and (IP) broadband backhaul and some Motorola phone. When you're in range of the basesation you roam on to it and calls will go that way (and at a cheaper rate). Though the specs are freely available, you need operator co-operation for it to happen, which is the stumbling block for most players. BT use Vodafone, though BT Mobile is a MVNO of Vodafone which probably helps, though they're also big enough to be very persuasive. Thanks for the info, if nothing else I'm now a little wiser on the subject. Wisdom is everything ;) Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING
On 3/18/06, Watkins, Bradley [EMAIL PROTECTED] wrote: cluster (or clusters, in the case of one site). So there is no NAT, and it is an Asterisk-only solution (at least insofar as telephony software is concerned). I'm just barging in.. This all looks 'very' promising stuff, I'm looking forward to any drafts/further discussions on the list. In the meanwhile it looks like I have to build some test-boxes to start trying this.. :) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Realtime Users
Douglas Garstang wrote:
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voxadmin for 6 seconds.
I can get information for the extension in question...
*CLI realtime load sipusers name 2944093
Column Name Column Value
id 1
name 2944093
accountcode 2944093
callgroup 1
canreinvite no
context
dtmfmode auto
nat rfc35
pickupgroup 1
qualify no
type friend
username 2944093
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
regseconds 0
cancallforward yes
subscribecontext sub_oneeighty
First of all, why doesn't Asterisk show _ALL_ the fields in the table? There's
way more than this.
Second, when my phone comes up, asterisk displays this on the console:
*CLI Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854 handle_request_register:
Registration from 'sip:[EMAIL PROTECTED]' failed for '216.xxx.142.205' -
Username/auth name mismatch
I'm trying to do this in insecure mode, so Asterisk shouldn't even be asking the phone for a password. What's the deal? When I run an ngrep on the database, I can see that Asterisk isn't even TRYING to query the extension. Huh??? My sip.conf just has a [global] section, no users are provisioned in it.
Doug.
Hi,
do you have in sip.conf
[From_OneEighty]
switch = Realtime/[EMAIL PROTECTED]
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Re: [Asterisk-Users] gsm picocells
On Friday 17 March 2006 23:23, James Harper wrote: Care to give me any more clues? Google only wants to tell me about articles about the use of picocells in aircraft and how much better the world will be when it happens :) Maybe I'm using the wrong search terms. I apologize; When I was googling for this about 4 months ago I was drowning in a sea of products. Now I can not find the ones I was trying to point you to. The closest link I have found is http://www.samsung.com/Products/WirelessSystems/CDMAInfrastructure/BaseTransceiverStation.asp Which describes a CDMA BTS, but not a micro/nano/pico one for use in buildings and so on. http://www.motorola.com/content/0,,5903-9039,00.html describes a Motorola GSM micro BTS, but that's still too large. I was *positive* that the ones I was looking at were from Samsung but their website has no mention of them anymore. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List of transcoding combinations
Is there a list or matrix somewhere that shows what codec can be transcoded? I am playing with different allowed codecs between my asterisk box and some of my providers testing voice quality and bandwidth usage on my cable connection, and I occassionally run into an issue where asterisk cannot convert between two codecs. For instance G.723 and ULAW will not work together through asterisk. Would like to have a matrix of some sort where I know ahead of time what combinations I can and cannot use. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List of transcoding combinations
Robert Webb wrote: Is there a list or matrix somewhere that shows what codec can be transcoded? I am playing with different allowed codecs between my asterisk box and some of my providers testing voice quality and bandwidth usage on my cable connection, and I occassionally run into an issue where asterisk cannot convert between two codecs. For instance G.723 and ULAW will not work together through asterisk. Would like to have a matrix of some sort where I know ahead of time what combinations I can and cannot use. Thanks Hi, you are going to laugh :) on cli show translation ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about meetme app
Thanks Jonathan. In this case, how do you actually mute everybody but the admins? Imagine giving a training to 100 people, and not wanting anybody to say anything except the trainer... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Augenstine Sent: March 17, 2006 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Question about meetme app My mistake. Locking a conference from the CLI does prevent any additional callers from connecting. But AFAIK locking the conference does not prevent you from muting a participant. What I was thinking in my original response was limiting a conference, not locking it, by adding a pin number. On Fri, 2006-03-17 at 17:45 -0500, Michael Gaudette wrote: As in press 2 to lock or unlock this conference in the conf admin menu? Then, how do you mute participants? I can't imagine MeetMe not having this functionality. Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Augenstine Sent: March 17, 2006 5:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about meetme app A locked conference means that a pin number is required to join the conference. On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote: I have a quick question about the MeetMe app. A locked conference means what exactly? A) That people can't join anymore B) That everyone is muted except the admin Follow-up question If the answer above is A, how do you accomplish B? Mick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog POTS line - Rhino FXO Channel Bank - No Hangup
Thanks for the link. The ultimate solution was to change from fxs_ls to fxs_ks. Now it works great! Thanks, James Dr. Michael J. Chudobiak wrote: [EMAIL PROTECTED] wrote: If so, is there a way to detect the hangup? Check out http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html for some possible clues. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Question about meetme app
In article [EMAIL PROTECTED],
Michael Gaudette [EMAIL PROTECTED] wrote:
Thanks Jonathan.
In this case, how do you actually mute everybody but the admins?
Imagine giving a training to 100 people, and not wanting anybody to say
anything except the trainer...
Here's an idea.
Have the leader enter MeetMe with the X option. In the same context, have
an extension number, say 1, which calls MeetMeAdmin with the N flag, which
means Mute All except Admins. It then puts him back into the MeetMe straight
away. You probably want to use the 'q' flag to suppress the enter and leave
sounds.
[meetme-admin]
exten = _X.,1,Set(CONF=${EXTEN})
exten = _X.,2,Answer
exten = _X.,3,MeetMe(${CONF}|daAqX)
exten = _X.,4,Hangup
exten = 1,1,MeetMeAdmin(${CONF}|N)
exten = 1,2,MeetMe(${CONF}|daAqX)
exten = 1,3,Hangup
exten = 2,1,MeetMeAdmin(${CONF}|n)
exten = 2,2,MeetMe(${CONF}|daAqX)
exten = 2,3,Hangup
[meetme-others]
exten = _X.,1,Set(CONF=${EXTEN})
exten = _X.,2,Answer
exten = _X.,3,MeetMe(${CONF}|dqwx)
exten = _X.,4,Hangup
So with the above, an admin ought to be able to press '1' to mute everyone
except the admins, and '2' to unmute them again.
Further functionality can be added using similar techniques.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] I have my asterisk machine behind a Linux, Nat ...
I would like to make a suggestion and recommend that you put your Asterisk box on the outside and let it also pull duty as your firewall/nat router. The iptables overhead will be minimal on the system and you'll save yourself a lot of headaches in the long run. The biggest problem being that having an asterisk server behind a nat, and then also having sip phones trying to connect to said server across the internet, which are most likely behind their own nats creates lots of issues. For instance you'll see that the phone registers with the server ok but cannot make calls, or you'll have one-way voice issues, etc, etc. If you need some help getting it set up this way contact me off-list and I'll give you a hand. I've done it several times this way and its not really that hard. Regards, Steve Cayona Date: Sat, 18 Mar 2006 08:43:00 +0100 From: Anthony Azzopardi [EMAIL PROTECTED] Subject: [Asterisk-Users] I have my asterisk machine behind a Linux Nat ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Hello ppl, I have my asterisk machine behind a Linux Nat router which is connected to the internet. Please tell me the iptables rules and other configurations that I need so that a sip phones on the internet can access asterisk. Best regards, Anthony. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More Voicemail prompts
Can Comedian Mail handle more than just an away and busy message? I've got a client that would like even more of them. I can write an app to replace messages externally, but I was wondering of comedian could handle it internally. As far as I know, no. But, what I did for a customer of mine is build a script that query a MySQL table to see if there is a special message to play for this particular date. Then I play that message with Playback and then send the call to voicemail with the option s. Have a look here : http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 DMTF
I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=inband context=pstn-in insecure=very -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Realtime Users
Yusuf,
No I don't have the switch statement in extensions.conf. I'm not trying to do
realtime extensions. I'm trying to do realtime SIP. They're different.
Doug.
-Original Message-
From: yusuf [mailto:[EMAIL PROTECTED]
Sent: Sat 3/18/2006 6:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] SIP Realtime Users
Douglas Garstang wrote:
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voxadmin for
6 seconds.
I can get information for the extension in question...
*CLI realtime load sipusers name 2944093
Column Name Column Value
id 1
name 2944093
accountcode 2944093
callgroup 1
canreinvite no
context
dtmfmode auto
nat rfc35
pickupgroup 1
qualify no
type friend
username 2944093
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
regseconds 0
cancallforward yes
subscribecontext sub_oneeighty
First of all, why doesn't Asterisk show _ALL_ the fields in the
table? There's way more than this.
Second, when my phone comes up, asterisk displays this on the console:
*CLI Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854
handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for
'216.xxx.142.205' - Username/auth name mismatch
I'm trying to do this in insecure mode, so Asterisk shouldn't even be
asking the phone for a password. What's the deal? When I run an ngrep on the
database, I can see that Asterisk isn't even TRYING to query the extension.
Huh??? My sip.conf just has a [global] section, no users are provisioned in it.
Doug.
Hi,
do you have in sip.conf
[From_OneEighty]
switch = Realtime/[EMAIL PROTECTED]
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RE : [Asterisk-Users] Sipura 3000 DMTF
Check for : dtmfmode=outband Good luck ! Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Chris Mason (Lists) Envoyé : samedi 18 mars 2006 17:43 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Sipura 3000 DMTF I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=inband context=pstn-in insecure=very -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 DMTF
Try with dtmfmode=auto and DTMF Tx Method: InBand+INFO, this was the best configuration for me, although still not 100% guarantee. If the dtmf tones are sent very fast without a 1 sec delay, in most of the cases asterisk won't detect half of them. There are a couple of patches for the trunk regarding this issue, but they didn't work for me. HTH, Vahan Chris Mason (Lists) wrote: I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? [snip] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to enable talking in chanspy while spying?
hello i want to spy on a chennel listen the voice conversation between two person. i also want talk to one of them but others will not listen my voice. how can i configure this using ChanSpy? thanks atik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not getting?
On Mar 16, 2006, at 12:36 PM, Martin Joseph wrote: So, I am answering my own post (bad form I know)... I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. Ok, I have given up on this. There seems to be some kind of deal breaker issue with the RFC 2833 support either on the wellgate 3701A or somewhere between it and asterisk. I also figured out that I don't need the gateway to use RFC2833 for me to achieve my goals. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took the 101 from my AG168V ATA's configuration screen, as I know that device seemed to work fine through the old HT-488 fxo(via rfc2833). I then changed my asterisk extensions for both the FXS and FXO on the wellgate to include dtmfmode=rfc2833. I now have changed the dtmfmode= in both the wellgate FXS and FXO SIP extensions to be inband. This seems to work fine. This has brought me to a point where both my hardphones (ATA's) and my softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail. Now I realize that Asterisk can handle my off site hardphones via RFC 2833 and my softphones via IAX2 just fine, and when the tones come from those sources(via RFC 2833) asterisk handles the conversion to inband and that works ok too. To me this means that asterisk is properly getting the RFC2833 events from the user agents. BUT, if I try to dial out the FXO, none of my phones (hard or soft) produce working touchtones for a PSTN based voicemail system. Even stranger to me, is the fact that from the phone connected to the FXS on the wellgate I can hear tones(listening on a called line), but they sound kind rough at the edges. From the AG168V I hear no tones, but what seems to be blown out tones (ie overdriven volume). From the IAX softphones I hear no tones at all just clicks! Now I would have guessed that the FXO would be doing the conversion of the RFC2833 to inband (for PSTN), so that I thought all the tones should sound the same from any phone? Apparently this isn't the case at all. Thanks to all of you for any help understanding and or debugging this mess. I would still love to hear an explanation for the variety of results I saw above? Even a broken wellgate RFC 2833 implementation doesn't seem to adequately explain it. I am sure it's my problem understanding how this works? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime SIP users/peers
Just spent hours dicking around with SIP Realtime. Every time a phone came up and sent a registration to Asterisk, Asterisk would simply NOT query the database. I had sipusers in extconfig, but added sippeers as well. NOW I can see Asterisk doing a 'SELECT * FROM sippeers WHERE name = '2944093''. Huh??? Uhm, why? It's not a peer! It's a bloody phone, and in my mind should be a user or a friend! It should be looking in sippeers! How does it decide which table to use? Has anyone made sense of this mess? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jittery meetme conference using Linksys 942 phones
We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values did not help. We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a TE-210 Dual-T1 card plugged in. The meetme.conf file has no general settings, just a list of two conference rooms. Has anyone else experienced sound quality issues with meetme conferences using Linksys phones? Any idea what could fix this? Thanks. Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime SIP users/peers - Screwed?
Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any records in sip_peers where the IP address of the message matches. What happens if this user may make calls from multiple IP addresses? Will I need one entry for each IP address that calls may come from? Will this even work? Would I be so frustrated if this stuff was documented somewhere? -Original Message- From: Douglas Garstang Sent: Saturday, March 18, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Realtime SIP users/peers Just spent hours dicking around with SIP Realtime. Every time a phone came up and sent a registration to Asterisk, Asterisk would simply NOT query the database. I had sipusers in extconfig, but added sippeers as well. NOW I can see Asterisk doing a 'SELECT * FROM sippeers WHERE name = '2944093''. Huh??? Uhm, why? It's not a peer! It's a bloody phone, and in my mind should be a user or a friend! It should be looking in sippeers! How does it decide which table to use? Has anyone made sense of this mess? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to enable talking in chanspy while spying?
This is an age old question. Unless something has changed, it is possible but not included functionality. A group of people paid to have this functionality developed but since they paid they decided not to release it back into the asterisk community. I am not sure if it for sale or not or even if it is, what the cost is. If you are listening to a zap channel with zapscan, it works like we want but not with chanspy (my understanding anyways). I need the same functionality for my call center so if you find a solution (even if it has to be purchased) please post back to the list. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of atik khan Sent: Saturday, March 18, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to enable talking in chanspy while spying? hello i want to spy on a chennel listen the voice conversation between two person. i also want talk to one of them but others will not listen my voice. how can i configure this using ChanSpy? thanks atik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] TDM 2400 With 24 FXO
Hello Fernando, I have checked this card with and without hardware echocan : the hardware echocan module does the job better than the zaptel software can do it. I recommand this module without any doubt. But, the echocan algorithms in zaptel are better and better and the CPUs power grows permanently. It is possible to use this card without hardware echocan, but you will encounter the same results, in this case, as you can obtain with the other TDM Digium's cards : correct for certain situations, not for all extreme cases, depending what listening level your users want, lines specifications and what critical echo threshold they can admit before to not be able to do correctly their job. Near same thing for E1/T1 harware echocan features. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Fernando BERRETTA Envoyé : vendredi 17 mars 2006 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] TDM 2400 With 24 FXO Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 DMTF
unsubscribe please. I tried the web site way, but doesn't seem to work. Thanks John B - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 9:17 AM Subject: Re: [Asterisk-Users] Sipura 3000 DMTF Chris, I've had my spa3k set to DTMF Tx Method: Auto and no dtmf entry in the sip.conf, and it seems to work. I don't use the spa3k much, but the tests that I did some time ago I believed worked just fine. Asterisk is running svn trunk from a few weeks ago. Rich Chris Mason (Lists) wrote: I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=inband context=pstn-in insecure=very ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1450 (20060318) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP users/peers - Screwed?
On 18 Mar 2006, at 19:21, Douglas Garstang wrote: Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any records in sip_peers where the IP address of the message matches. What happens if this user may make calls from multiple IP addresses? Will I need one entry for each IP address that calls may come from? Will this even work? Would I be so frustrated if this stuff was documented somewhere? From the wiki: --- Asterisk matches incoming calls to the name of a device with type=user based on the From: user name (ignoring the SIP domain). The other way that incoming SIP requests are matched to [xxx] sections in this file, is to examine the IP address that the request is coming from, and look for a peer [xxx] section that has a matching Host= value. If Host=dynamic, then no match is possible until the SIP client has registered. and - When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer] section matching the caller's IP address. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. -- (I know that doesn't entirely explain the behavior but it is a start) I'm guessing that the sql query immediately before your extract was a name search that came up with nothing Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P, with Tzafir, Anton, Cosmin, Colin...
Hi Yrving. I dont use [EMAIL PROTECTED] but if you ever use plain ol' asterisk, I might be able to give you a hand. drop me aline when you do. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yrving rivasSent: Tuesday, March 07, 2006 6:18 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] fax receive using TDM400P, with Tzafir, Anton,Cosmin, Colin... Dear friends: I have seen Tzafir, Anton, Cosmin, Colin and other very interesting peopleworking very hard with the fax, almost at the point to write a book (I hope some day they will for all of us). I have been reading and saving all of those mails carefully to find the key to my needs. But their knowledge is too high for me. Can any of you explain me, send me a document or refer me to a book where I can find step by step (for a person like me who doesn´t know linux more than a couple of commands) how to make the faxwork? My [EMAIL PROTECTED] with tdm400p w/4 fxo ports, seems to negotiate...but I don´t know where to find the files. Before I used to go to my webmail in de AMP and see some of the files there, and when I opened them, all pages where whith nothing in. What I want is the [EMAIL PROTECTED] to receive my faxes and then send it to my email. Thanks. Yrving Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.com.mx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom voice.gain.tx.analog.handsetandasteriskecho
Man! I love polycoms.. They are good phones and highly configurable. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Tuesday, March 07, 2006 7:41 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Polycom |voice.gain.tx.analog.handsetandasteriskecho | ||While I'm asking about the Polycom ip500, the answers for all phones ||where mic/handset/headset levels are adjustable would be of |interest to ||many I'm sure. || ||For the ip500, the default value for the handset seems to be ||voice.gain.tx.analog.handset=3 | |I have a number of IP600s and 601s that I was experiencing |occassional echo with. I recently upgraded them to firmware |1.6.5, and rather than using my existing sip.cfg/ipmid.cfg |that had been around forever I started fresh with a completely |stock 1.6.5 sip.cfg file. My echo issues have disappeared completely. | |With the 1.6.5 version of the Polycom firmware the default |value for voice.gain.tx.analog.handset=12. The default |value for voice.gain.tx.analog.headset=3. I suspect you |should update the entire voice section of the file (if |you're not ready to start from scratch) since it contains |default values for AEC, AES, NS, AGC, RXEQ, and TXEQ. I have |pasted just the gains section below in case anyone want to |compare it to their current settings. | | gains | voice.gain.rx.analog.handset=0 | voice.gain.rx.analog.headset=0 | voice.gain.rx.analog.chassis=0 | voice.gain.rx.analog.chassis.IP_300=-6 | voice.gain.rx.analog.chassis.IP_4000=3 | voice.gain.rx.analog.chassis.IP_601=6 | voice.gain.rx.analog.ringer=0 | voice.gain.rx.analog.ringer.IP_300=-6 | voice.gain.rx.analog.ringer.IP_4000=3 | voice.gain.rx.analog.ringer.IP_601=6 | voice.gain.rx.digital.handset=-15 | voice.gain.rx.digital.headset=-21 | voice.gain.rx.digital.chassis=0 | voice.gain.rx.digital.chassis.IP_4000=0 | voice.gain.rx.digital.chassis.IP_601=0 | voice.gain.rx.digital.ringer=-21 | voice.gain.rx.digital.ringer.IP_4000=-21 | voice.gain.rx.digital.ringer.IP_601=-21 | voice.gain.rx.analog.handset.sidetone=-14 | voice.gain.rx.analog.headset.sidetone=-24 | voice.gain.tx.analog.handset=12 | voice.gain.tx.analog.headset=3 | voice.gain.tx.analog.chassis=3 | voice.gain.tx.analog.chassis.IP_300=0 | voice.gain.tx.analog.chassis.IP_4000=3 | voice.gain.tx.analog.chassis.IP_601=0 | voice.gain.tx.digital.handset=0 | voice.gain.tx.digital.headset=0 | voice.gain.tx.digital.chassis=3 | voice.gain.tx.digital.chassis.IP_4000=0 | voice.gain.tx.digital.chassis.IP_601=6 | voice.gain.tx.analog.preamp.handset=14 | voice.gain.tx.analog.preamp.headset=23 | voice.gain.tx.analog.preamp.chassis=32 | voice.gain.tx.analog.preamp.chassis.IP_601=32/ | |-- E | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Panasonic KX-TDA1000 with asterisk server
Has anyone have integrated Panasonic PBX QSIG with asterisk servers using E1 interfase? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Jittery meetme conference using Linksys 942 phones
I have not used the Linksys 942 phones yet, but I have a couple of Sipura 841's. Check to see what your RTP payload encoding frame length is, ie. 20ms or 30ms. Also check to see if there is a setting to surpress or transmit silence. If so you want to transmit silence. Rana Dutt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values did not help. We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a TE-210 Dual-T1 card plugged in. The meetme.conf file has no general settings, just a list of two conference rooms. Has anyone else experienced sound quality issues with meetme conferences using Linksys phones? Any idea what could fix this? Thanks. Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
On Mar 18, 2006, at 2:05 PM, [EMAIL PROTECTED] wrote: I was also thinking a list for newbies... As a newb I think that is a bad idea. First of all, the heavy hitters will all want to avoid it( a newb list). Secondly I have learned a LOT just by reading other peoples (non newbs) problems and solutions... It's true there is a lot of traffic here, but personally I prefer too many messages to too few... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
Hi all, First post, etc, etc. :) Robert La Ferla wrote: Perhaps, we could split the list into two: As the list uses Mailman, it is possible to specify topics on the subject prefix and then set your Mailman preferences to select which topics you like. The admin could, for a simple example, have NEWBIES:, SOFTWARE:, HARDWARE:, UBERGEEK:, PHONES: and MISC: topics and then set the preferences to require a topic (invalid/missing topic = bounce). Then people can pick the topics they like and Robert's your mother's brother. It's just an idea I have from another Mailman-based list I'm on where it's worked very well for years. This is of course ignoring the fact there are numerous Asterisk lists on this server already. :) Cheers, Matt While money can't buy happiness, it certainly lets you choose your own form of misery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test
HI, all This is a test. By some reason I stopped received e-mails from the list. Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on hosted server
On Friday March 17 2006 8:07 am, Can2002 wrote: I'd planning on install Asterisk on a hosted Linux box we're setting up. The hosting provider that seems to offer the best deal can install either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions (running on AMD 64 bit). My experience has been gained on RedHat to date, but I do have some SUSE experience. I assume I'll have no problems running Asterisk on SUSE, but I'd appreciate any recommendations. Also, should I be safe on a 64 bit distribution? Cheers, Chris Chris, I have been away for a while so not sure if you have gotten many answers on this yet but... I am running * 1.2.4 on an AMD Opteron 165 dual core with SUSE 10.x 64 bit and all is working very well. John M ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jittery meetme conference using Linksys 942 phones
What are your zttest results? zttest can be run from /usr/src/zaptel/ directory (run ./zttest from there). Do you have Digium hardware or ztdummy? Pedro http://www.TRACI.netOn 3/18/06, Rana Dutt [EMAIL PROTECTED] wrote: We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values did not help. We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a TE-210 Dual-T1 card plugged in. The meetme.conf file has no general settings, just a list of two conference rooms. Has anyone else experienced sound quality issues with meetme conferences using Linksys phones? Any idea what could fix this? Thanks. Ron ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 dual ethernet port - bandwidth impact
FYI for anyone using the dual ethernet ports on a Cisco 7960. I'm using a Cisco 7960 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - 7960 - HP Switch - Asterisk : actual measured at 93.02 mbps. The second test (through the 7960) was conducted with a g711 conversation in progress with absolutely no noticeable impact to the audio quality. The 7960 was running sip v7.1 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!!
Seems to me that it's more logical for the phones to know what their SRV records are than the server. You shouldn't rely on the dns to ensure that your system is redundant. Aaron On Mar 16, 2006, at 1:03 PM, Douglas Garstang wrote: I know someone who's at VON this week. Apparently Mark Spencer was up there talking about how Asterisk supports SRV. Sounds like vaporware to me. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Thursday, March 16, 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! In regards to HA... SER is definitely a good option, but it does require the extra hardware to have at least 2 boxes that can failover on each other. I would user OpenSER however (better documentation and mor features). I couldn't agree more that Asterisk should FULLY support DNS-SRV. The solution seems to work great for phones and ATA's. This would be a good item to create a bounty for. I have only two boxes right now, so it seems like my only HA options are the dreaded DUNDi setup or a active/passive failover with linux-HA. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GS BT102 dual ethernet port -bandwidth impact
FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - BT102 - HP Switch - Asterisk : actual measured at 8.86 mbps. The second test (through the BT102) was conducted with a g711 conversation in progress. Audio quality was noticeably impacted presumably due to the half duplex support in the BT102. The BT102 was running sip v1.0.5.18 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. The tests were run purely to document throughput of the phone when used with an attached PC. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP600 dual ethernet port - bandwidth impact
FYI for anyone using the dual ethernet ports on a Polycom IP600. I'm using a Polycom IP600 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps. PC - IP600 - HP Switch - Asterisk : actual measured at 91.9 mbps. The second test (through the IP600) was conducted with a g711 conversation in progress with absolutely no noticeable impact to the audio quality. The IP600 was running sip v1.5.2.0054 firmware. The bandwidth tester (older version of NetIQ's QCheck) sent one megabyte bursts of tcp traffic between the two endpoints using 1514 bytes packets. The tests were run purely to document throughput of the phone when used with an attached PC. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
Splitting the list by type of request may be a good idea, but splitting based on skill level is just a bad idea... I'm pretty sure that regardless of a newbie's status, they'll still just go to the other lists as the newbie list likely won't do much good. In short, I agree with different lists for hardware and configuration questions... Aaron On Mar 18, 2006, at 4:05 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
This same issue has been discussed many times over the last two years. Not likely its going to change now. Aaron Daniel wrote: Splitting the list by type of request may be a good idea, but splitting based on skill level is just a bad idea... I'm pretty sure that regardless of a newbie's status, they'll still just go to the other lists as the newbie list likely won't do much good. In short, I agree with different lists for hardware and configuration questions... Aaron On Mar 18, 2006, at 4:05 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP600 - no ring?
Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The ip600 is running v1.5.2.0054 and works very well in all other respects. Any thoughts on how to diagnose this? (Its definitely not an asterisk config problem.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A general deployment question (OT)
Does anyone have a guesstimate of how many active Asterisk installations there are? Sorry this is off topic, need it for a customer proposal and they need comfort on stability. A count of the downloads from Digium would be a good start but I couldn't find this anywhere with Google. Feedback from anyone who may know near actual data would be appreciated rather than simply guessing, as I hope this doesn't generate too many posts (sorry if it does). Cheers Rob ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP600 - no ring?
Have a look in the Polycom phone directory - see if the number of the first 7960 is defined in there with a ring type of 0 (silent). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, 19 March 2006 1:51 PM To: Asterisk Users-List Subject: [Asterisk-Users] Polycom IP600 - no ring? Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The ip600 is running v1.5.2.0054 and works very well in all other respects. Any thoughts on how to diagnose this? (Its definitely not an asterisk config problem.) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!!
Huh? Phones do a NAPTR/SRV lookup in a specified domain to get a list of SRV records to use. The phones don't query the DNS server every time they make a call... they have a cache. You also run primary and a secondary (or two primary) dns servers. It's a simple scalable solution. It's a shame Asterisk doesn't support it. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Sat 3/18/2006 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA andPolycomphone!! Seems to me that it's more logical for the phones to know what their SRV records are than the server. You shouldn't rely on the dns to ensure that your system is redundant. Aaron On Mar 16, 2006, at 1:03 PM, Douglas Garstang wrote: I know someone who's at VON this week. Apparently Mark Spencer was up there talking about how Asterisk supports SRV. Sounds like vaporware to me. -Original Message- From: David Thomas [mailto:[EMAIL PROTECTED] Sent: Thursday, March 16, 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA and Polycomphone!! In regards to HA... SER is definitely a good option, but it does require the extra hardware to have at least 2 boxes that can failover on each other. I would user OpenSER however (better documentation and mor features). I couldn't agree more that Asterisk should FULLY support DNS-SRV. The solution seems to work great for phones and ATA's. This would be a good item to create a bounty for. I have only two boxes right now, so it seems like my only HA options are the dreaded DUNDi setup or a active/passive failover with linux-HA. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Asterisk Users Mailing List Traffic
Of course, but if newbies are separated and together only without any expert, who can explain them anything ? I am actualy a subscriber for all the Digium lists. If more lists will be, more subscribtions I will get and I will receive the same quantity of messages ;-) Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : samedi 18 mars 2006 23:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk Users Mailing List Traffic I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 and latency measures
Hi, we have set up a small project in a school the following way: SITE_A(4 port analog to ip g729)--ADSL_ISP1---ISP2Asterisk-PSTN Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit) The asterisk box gets internet service via a wireless antenna. 1 Mbit of up/down bandwith Comments: So far, this means that I will need licenses for the 729. asterisk only supports 20ms sampling on g729 so 4 channels will need 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet bandwith. i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware. Installing ADSL from ISP1 on the asterisk place will give a clear advantage Please correct any of my prior statements if wrong. should I maintain packet latency below 300ms or 150ms? How can I measure this latency all the way to the asterisk? Should I ping from SITE_A to the asterisk box with 8k packets? If I can't install ADSL for the moment, will the above setup work? thanks in advance for all your help. Erick. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] An FXO version of IAXy?
Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to enable talking in chanspy while spying?
Does anyone know how much was paid ? We would be willing to part-fund this and to release it as part of the distribution. Julian. Steven Totaro wrote: This is an age old question. Unless something has changed, it is possible but not included functionality. A group of people paid to have this functionality developed but since they paid they decided not to release it back into the asterisk community. I am not sure if it for sale or not or even if it is, what the cost is. If you are listening to a zap channel with zapscan, it works like we want but not with chanspy (my understanding anyways). I need the same functionality for my call center so if you find a solution (even if it has to be purchased) please post back to the list. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of atik khan Sent: Saturday, March 18, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to enable talking in chanspy while spying? hello i want to spy on a chennel listen the voice conversation between two person. i also want talk to one of them but others will not listen my voice. how can i configure this using ChanSpy? thanks atik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] An FXO version of IAXy?
On Mar 18, 2006, at 11:31 PM, Steve Murphy wrote: Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? Yes! I think a 1 port FXO and/or 2 port FXO/FXS IAXy would be a great addition. I have been in search of a device like this for about 4 months now, and neither of the devices I have tried (HT-488 and Wellgate 3701a) have really been particularly well thought out (UNDERSTATEMENT). I think for the SOHO market and also for people who are just learning about asterisk, such an item would be a great addition to digiums Product portfolio. I certainly would love to support Digium by buying such an item. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
